Adtran T908

I have a customer that has a PRI that goes into a Adtran T908 to convert it to SIP for the current phone system (not a freepbx). I am trying to add the PBX to the Adtran so it can make outgoing calls and will eventually use the Adtran as the SIP Gateway for the new PBX, but now I am just testing. For the life of me I cannot get a outgoing call to work. The Adtran always responds 404 not found to all call invites. Has anyone been able to get a Adtran T908 with a PRI setup to talk to Freepbx? When I look at the wireshark of a successful call with the old pbx and compare it with a wireshark from the freepbx, I cannot see a difference in the headers, they are in the same format. Any Suggestions?

Adtran will respond 404 if the incoming invite is not recognized as being from a valid trunk.

Make sure you have set up your PBX as a trunk on the Adtran. If you are using the CLI it can be as simple as this:

voice trunk T01 type sip
  sip-server primary IPADDRESS
  description "FreePBX"

It is - (I did set “From Domain” in PJSIP to match “grammar from host local”. I had to, for some reason freepbx wants to put the WAN address in there by default. )

voice trunk T04 type sip
description "Sangoma"
sip-server primary
dial-string source to
grammar from host local
transfer-mode network

This should not happen if you include the subnet of the Adtran in the Local Networks field under Asterisk SIP Settings.

On the Adtran, you can get useful debug information with debug sip stack messages to see the SIP as the Adtran sees it, and debug voice summary or verbose to see call routing logic.

Yes I agree with you on the freepbx address. Makes no sense to me as all the rest of the headers and fields have the local address, it was just the from header that had the WAN in it. SIP settings is correct and I rebooted the entire system and it sill insisted on putting the WAN address in the from header, contact header was correct.

Thank you for the “voice verbose” suggestion, I was looking at the sip and sip stack but I did not think of the voice debug. I found it! It said it was denied access to the trunk. After some digging I found there was a trunk group restriction on the isdn trunk that limited access to the sip trunks to the old pbx, so I added the new trunk to it and now works. Thanks!!


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